Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h |
index f59676685dbe81e27843953680f3b21ef155ff02..9c2a71499ad88152e3f9c18434387df98f34d413 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h |
@@ -21,6 +21,7 @@ |
#include "webrtc/api/call/transport.h" |
#include "webrtc/base/constructormagic.h" |
#include "webrtc/base/criticalsection.h" |
+#include "webrtc/base/optional.h" |
#include "webrtc/base/random.h" |
#include "webrtc/base/thread_annotations.h" |
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" |
@@ -150,6 +151,7 @@ class RTCPSender { |
void SetCsrcs(const std::vector<uint32_t>& csrcs); |
void SetTargetBitrate(unsigned int target_bitrate); |
+ void SetVideoBitrateAllocation(const BitrateAllocation& bitrate); |
bool SendFeedbackPacket(const rtcp::TransportFeedback& packet); |
private: |
@@ -197,6 +199,9 @@ class RTCPSender { |
EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
std::unique_ptr<rtcp::RtcpPacket> BuildDlrr(const RtcpContext& context) |
EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
+ std::unique_ptr<rtcp::RtcpPacket> BuildTargetBitrate( |
+ const RtcpContext& context) |
+ EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
private: |
const bool audio_; |
@@ -265,6 +270,9 @@ class RTCPSender { |
RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_); |
+ rtc::Optional<BitrateAllocation> video_bitrate_allocation_ |
+ GUARDED_BY(critical_section_rtcp_sender_); |
+ |
void SetFlag(RTCPPacketType type, bool is_volatile) |
EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile) |